Searched refs:webrtc (Results 1 - 25 of 493) sorted by relevance

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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/
H A DRTCEnumConverter.mm30 #include "talk/app/webrtc/peerconnectioninterface.h"
35 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
37 case webrtc::PeerConnectionInterface::kIceConnectionNew:
39 case webrtc::PeerConnectionInterface::kIceConnectionChecking:
41 case webrtc::PeerConnectionInterface::kIceConnectionConnected:
43 case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
45 case webrtc::PeerConnectionInterface::kIceConnectionFailed:
47 case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
49 case webrtc::PeerConnectionInterface::kIceConnectionClosed:
55 (webrtc
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H A DRTCSessionDescription+Internal.h30 #include "talk/app/webrtc/jsep.h"
31 #include "talk/app/webrtc/webrtcsession.h"
36 - (webrtc::SessionDescriptionInterface *)sessionDescription;
39 (const webrtc::SessionDescriptionInterface*)sessionDescription;
H A DRTCICECandidate+Internal.h30 #include "talk/app/webrtc/peerconnectioninterface.h"
35 webrtc::IceCandidateInterface* candidate;
37 - (id)initWithCandidate:(const webrtc::IceCandidateInterface*)candidate;
H A DRTCVideoRenderer+Internal.h30 #include "talk/app/webrtc/mediastreaminterface.h"
36 webrtc::VideoRendererInterface *videoRenderer;
38 - (id)initWithVideoRenderer:(webrtc::VideoRendererInterface *)videoRenderer;
H A DRTCMediaConstraints.mm42 talk_base::scoped_ptr<webrtc::RTCMediaConstraintsNative> _constraints;
43 webrtc::MediaConstraintsInterface::Constraints _mandatory;
44 webrtc::MediaConstraintsInterface::Constraints _optional;
53 new webrtc::RTCMediaConstraintsNative(_mandatory, _optional));
58 + (webrtc::MediaConstraintsInterface::Constraints)
60 webrtc::MediaConstraintsInterface::Constraints constraints;
62 constraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
72 - (const webrtc::RTCMediaConstraintsNative *)constraints {
H A DRTCEnumConverter.h32 #include "talk/app/webrtc/peerconnectioninterface.h"
37 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
40 (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
43 (webrtc::PeerConnectionInterface::SignalingState)nativeState;
46 (webrtc::MediaSourceInterface::SourceState)nativeState;
48 + (webrtc::MediaStreamTrackInterface::TrackState)convertTrackStateToNative:
52 (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
H A DRTCPeerConnection+Internal.h33 #include "talk/app/webrtc/peerconnectioninterface.h"
38 talk_base::scoped_refptr<webrtc::PeerConnectionInterface> peerConnection;
41 talk_base::scoped_refptr<webrtc::PeerConnectionInterface>)peerConnection
42 observer:(webrtc::RTCPeerConnectionObserver *)observer;
/external/chromium_org/remoting/host/
H A Dscreen_resolution.h10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
18 ScreenResolution(const webrtc::DesktopSize& dimensions,
19 const webrtc::DesktopVector& dpi);
22 webrtc::DesktopSize ScaleDimensionsToDpi(
23 const webrtc::DesktopVector& new_dpi) const;
26 const webrtc::DesktopSize& dimensions() const { return dimensions_; }
29 const webrtc::DesktopVector& dpi() const { return dpi_; }
36 webrtc::DesktopSize dimensions_;
37 webrtc::DesktopVector dpi_;
H A Dscreen_resolution_unittest.cc16 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10));
20 webrtc::DesktopSize(), webrtc::DesktopVector(10, 10));
24 webrtc::DesktopSize(1, 1), webrtc::DesktopVector(0, 0));
30 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10));
32 EXPECT_TRUE(webrtc::DesktopSize(50, 50).equals(
33 resolution.ScaleDimensionsToDpi(webrtc
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H A Dchromoting_param_traits.cc12 void ParamTraits<webrtc::DesktopVector>::Write(Message* m,
13 const webrtc::DesktopVector& p) {
19 bool ParamTraits<webrtc::DesktopVector>::Read(const Message* m,
21 webrtc::DesktopVector* r) {
25 *r = webrtc::DesktopVector(x, y);
30 void ParamTraits<webrtc::DesktopVector>::Log(const webrtc::DesktopVector& p,
32 l->append(base::StringPrintf("webrtc::DesktopVector(%d, %d)",
37 void ParamTraits<webrtc::DesktopSize>::Write(Message* m,
38 const webrtc
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
H A Dwebrtcvoe.h33 #include "talk/media/webrtc/webrtccommon.h"
35 #include "webrtc/common_types.h"
36 #include "webrtc/modules/audio_device/include/audio_device.h"
37 #include "webrtc/voice_engine/include/voe_audio_processing.h"
38 #include "webrtc/voice_engine/include/voe_base.h"
39 #include "webrtc/voice_engine/include/voe_codec.h"
40 #include "webrtc/voice_engine/include/voe_dtmf.h"
41 #include "webrtc/voice_engine/include/voe_errors.h"
42 #include "webrtc/voice_engine/include/voe_external_media.h"
43 #include "webrtc/voice_engin
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H A Dwebrtcvideodecoderfactory.h32 #include "webrtc/common_types.h"
34 namespace webrtc { namespace
44 virtual webrtc::VideoDecoder* CreateVideoDecoder(
45 webrtc::VideoCodecType type) = 0;
48 virtual void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) = 0;
H A Dwebrtcvie.h33 #include "talk/media/webrtc/webrtccommon.h"
34 #include "webrtc/common_types.h"
35 #include "webrtc/modules/interface/module_common_types.h"
36 #include "webrtc/modules/video_capture/include/video_capture.h"
37 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
38 #include "webrtc/modules/video_render/include/video_render.h"
39 #include "webrtc/video_engine/include/vie_base.h"
40 #include "webrtc/video_engine/include/vie_capture.h"
41 #include "webrtc/video_engine/include/vie_codec.h"
42 #include "webrtc/video_engin
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H A Dwebrtcvideocapturer.h36 #include "talk/media/webrtc/webrtcvideoframe.h"
37 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
38 #include "webrtc/modules/video_capture/include/video_capture.h"
47 virtual webrtc::VideoCaptureModule* Create(
49 virtual webrtc::VideoCaptureModule::DeviceInfo* CreateDeviceInfo(int id) = 0;
51 webrtc::VideoCaptureModule::DeviceInfo* info) = 0;
56 public webrtc::VideoCaptureDataCallback {
63 bool Init(webrtc::VideoCaptureModule* module);
80 webrtc::I420VideoFrame& frame);
82 webrtc
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/
H A Dfakemediastreamsignaling.h32 #include "talk/app/webrtc/audiotrack.h"
33 #include "talk/app/webrtc/mediastreamsignaling.h"
34 #include "talk/app/webrtc/videotrack.h"
44 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
45 public webrtc::MediaStreamSignalingObserver {
48 webrtc::MediaStreamSignaling(talk_base::Thread::Current(), this) {
88 virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {
90 virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {
92 virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {
94 virtual void OnAddLocalAudioTrack(webrtc
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H A Dtestsdpstrings.h33 namespace webrtc { namespace
80 "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"
142 } // namespace webrtc
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/
H A DAudioSource.java28 package org.webrtc;
H A DAudioTrack.java28 package org.webrtc;
H A DStatsObserver.java28 package org.webrtc;
30 /** Interface for observing Stats reports (see webrtc::StatsObservers). */
H A DVideoSource.java29 package org.webrtc;
/external/webrtc/src/system_wrappers/interface/
H A Dsleep.h15 namespace webrtc { namespace
22 } // namespace webrtc
/external/webrtc/src/system_wrappers/source/
H A Dcpu_no_op.cc15 namespace webrtc { namespace
22 } // namespace webrtc
H A Dcritical_section.cc18 namespace webrtc { namespace
27 } // namespace webrtc
H A Dtrace_unittest.cc17 using webrtc::CpuMeasurementHarness;
18 using webrtc::Trace;
19 using webrtc::kTraceWarning;
20 using webrtc::kTraceUtility;
22 class Logger : public webrtc::CpuTarget {
26 std::string trace_file = webrtc::test::OutputPath() +
29 Trace::SetLevelFilter(webrtc::kTraceAll);
/external/chromium_org/content/renderer/media/
H A Dmock_peer_connection_impl.h14 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
21 class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface {
26 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
28 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
31 webrtc::MediaStreamInterface* local_stream,
32 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
34 webrtc::MediaStreamInterface* local_stream) OVERRIDE;
35 virtual talk_base::scoped_refptr<webrtc::DtmfSenderInterface>
36 CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE;
37 virtual talk_base::scoped_refptr<webrtc
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